audio gaps, RTCP, TURN

[Daniel Pocock]

Hi all,

I've been testing Jami with somebody and we notice gaps in the audio.
We are both in Europe.

How does the RTP stack in Jami compare to other media stacks such as WebRTC?

Can I get RTCP statistics anywhere?

Could this be a problem with turn.jami.net?  I have my own TURN servers
and I could use them instead.

Is there any way I can remotely assist the other person to change their
TURN server?

Alternatively, if I configure my local Jami to use my TURN server, can I
block the peer's TURN server from being a candidate?

The audio is using Pulseaudio

The Audio codec Opus is enabled, all other codecs disabled

We did not try video

Regards,

Daniel